author | Sam Lantinga <slouken@libsdl.org> |
Mon, 10 Jul 2006 21:04:37 +0000 | |
changeset 1895 | c121d94672cb |
parent 1358 | c71e05b4dc2e |
child 1964 | 071b6598d48f |
permissions | -rw-r--r-- |
0 | 1 |
/* |
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SDL - Simple DirectMedia Layer |
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Copyright (C) 1997-2006 Sam Lantinga |
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This library is free software; you can redistribute it and/or |
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modify it under the terms of the GNU Lesser General Public |
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License as published by the Free Software Foundation; either |
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version 2.1 of the License, or (at your option) any later version. |
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This library is distributed in the hope that it will be useful, |
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but WITHOUT ANY WARRANTY; without even the implied warranty of |
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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Lesser General Public License for more details. |
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You should have received a copy of the GNU Lesser General Public |
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License along with this library; if not, write to the Free Software |
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Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA |
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Sam Lantinga |
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slouken@libsdl.org |
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*/ |
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/** |
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* \file SDL_audio.h |
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* |
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* Access to the raw audio mixing buffer for the SDL library |
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*/ |
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#ifndef _SDL_audio_h |
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#define _SDL_audio_h |
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#include "SDL_stdinc.h" |
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#include "SDL_error.h" |
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#include "SDL_endian.h" |
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#include "SDL_mutex.h" |
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#include "SDL_thread.h" |
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#include "SDL_rwops.h" |
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#include "begin_code.h" |
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/* Set up for C function definitions, even when using C++ */ |
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#ifdef __cplusplus |
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/* *INDENT-OFF* */ |
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extern "C" { |
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/* *INDENT-ON* */ |
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#endif |
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/* The calculated values in this structure are calculated by SDL_OpenAudio() */ |
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typedef struct SDL_AudioSpec |
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{ |
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int freq; /* DSP frequency -- samples per second */ |
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Uint16 format; /* Audio data format */ |
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Uint8 channels; /* Number of channels: 1 mono, 2 stereo */ |
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Uint8 silence; /* Audio buffer silence value (calculated) */ |
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Uint16 samples; /* Audio buffer size in samples (power of 2) */ |
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Uint16 padding; /* Necessary for some compile environments */ |
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Uint32 size; /* Audio buffer size in bytes (calculated) */ |
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/* This function is called when the audio device needs more data. |
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'stream' is a pointer to the audio data buffer |
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'len' is the length of that buffer in bytes. |
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Once the callback returns, the buffer will no longer be valid. |
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Stereo samples are stored in a LRLRLR ordering. |
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*/ |
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void (SDLCALL * callback) (void *userdata, Uint8 * stream, int len); |
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void *userdata; |
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} SDL_AudioSpec; |
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/* Audio format flags (defaults to LSB byte order) */ |
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#define AUDIO_U8 0x0008 /* Unsigned 8-bit samples */ |
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#define AUDIO_S8 0x8008 /* Signed 8-bit samples */ |
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#define AUDIO_U16LSB 0x0010 /* Unsigned 16-bit samples */ |
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#define AUDIO_S16LSB 0x8010 /* Signed 16-bit samples */ |
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#define AUDIO_U16MSB 0x1010 /* As above, but big-endian byte order */ |
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#define AUDIO_S16MSB 0x9010 /* As above, but big-endian byte order */ |
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#define AUDIO_U16 AUDIO_U16LSB |
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#define AUDIO_S16 AUDIO_S16LSB |
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/* Native audio byte ordering */ |
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#if SDL_BYTEORDER == SDL_LIL_ENDIAN |
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#define AUDIO_U16SYS AUDIO_U16LSB |
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#define AUDIO_S16SYS AUDIO_S16LSB |
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#else |
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#define AUDIO_U16SYS AUDIO_U16MSB |
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#define AUDIO_S16SYS AUDIO_S16MSB |
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#endif |
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/* A structure to hold a set of audio conversion filters and buffers */ |
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typedef struct SDL_AudioCVT |
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{ |
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int needed; /* Set to 1 if conversion possible */ |
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Uint16 src_format; /* Source audio format */ |
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Uint16 dst_format; /* Target audio format */ |
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double rate_incr; /* Rate conversion increment */ |
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Uint8 *buf; /* Buffer to hold entire audio data */ |
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int len; /* Length of original audio buffer */ |
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int len_cvt; /* Length of converted audio buffer */ |
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int len_mult; /* buffer must be len*len_mult big */ |
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double len_ratio; /* Given len, final size is len*len_ratio */ |
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void (SDLCALL * filters[10]) (struct SDL_AudioCVT * cvt, Uint16 format); |
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int filter_index; /* Current audio conversion function */ |
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} SDL_AudioCVT; |
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/* Function prototypes */ |
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/* These functions return the list of built in video drivers, in the |
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* order that they are normally initialized by default. |
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*/ |
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extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); |
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extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); |
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/* These functions are used internally, and should not be used unless you |
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* have a specific need to specify the audio driver you want to use. |
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* You should normally use SDL_Init() or SDL_InitSubSystem(). |
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*/ |
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extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); |
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extern DECLSPEC void SDLCALL SDL_AudioQuit(void); |
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/* This function returns the name of the current audio driver, or NULL |
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* if no driver has been initialized. |
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*/ |
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extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); |
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/* |
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* This function opens the audio device with the desired parameters, and |
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* returns 0 if successful, placing the actual hardware parameters in the |
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* structure pointed to by 'obtained'. If 'obtained' is NULL, the audio |
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* data passed to the callback function will be guaranteed to be in the |
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* requested format, and will be automatically converted to the hardware |
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* audio format if necessary. This function returns -1 if it failed |
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* to open the audio device, or couldn't set up the audio thread. |
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* |
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* When filling in the desired audio spec structure, |
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* 'desired->freq' should be the desired audio frequency in samples-per-second. |
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* 'desired->format' should be the desired audio format. |
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* 'desired->samples' is the desired size of the audio buffer, in samples. |
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* This number should be a power of two, and may be adjusted by the audio |
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* driver to a value more suitable for the hardware. Good values seem to |
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* range between 512 and 8096 inclusive, depending on the application and |
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* CPU speed. Smaller values yield faster response time, but can lead |
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* to underflow if the application is doing heavy processing and cannot |
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* fill the audio buffer in time. A stereo sample consists of both right |
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* and left channels in LR ordering. |
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* Note that the number of samples is directly related to time by the |
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* following formula: ms = (samples*1000)/freq |
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* 'desired->size' is the size in bytes of the audio buffer, and is |
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* calculated by SDL_OpenAudio(). |
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* 'desired->silence' is the value used to set the buffer to silence, |
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* and is calculated by SDL_OpenAudio(). |
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* 'desired->callback' should be set to a function that will be called |
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* when the audio device is ready for more data. It is passed a pointer |
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* to the audio buffer, and the length in bytes of the audio buffer. |
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* This function usually runs in a separate thread, and so you should |
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* protect data structures that it accesses by calling SDL_LockAudio() |
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* and SDL_UnlockAudio() in your code. |
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* 'desired->userdata' is passed as the first parameter to your callback |
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* function. |
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* |
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* The audio device starts out playing silence when it's opened, and should |
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* be enabled for playing by calling SDL_PauseAudio(0) when you are ready |
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* for your audio callback function to be called. Since the audio driver |
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* may modify the requested size of the audio buffer, you should allocate |
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* any local mixing buffers after you open the audio device. |
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*/ |
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extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, |
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SDL_AudioSpec * obtained); |
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/* |
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* Get the current audio state: |
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*/ |
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typedef enum |
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{ |
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SDL_AUDIO_STOPPED = 0, |
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SDL_AUDIO_PLAYING, |
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SDL_AUDIO_PAUSED |
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} SDL_audiostatus; |
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extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); |
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/* |
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* This function pauses and unpauses the audio callback processing. |
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* It should be called with a parameter of 0 after opening the audio |
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* device to start playing sound. This is so you can safely initialize |
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* data for your callback function after opening the audio device. |
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* Silence will be written to the audio device during the pause. |
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*/ |
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extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
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/* |
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* This function loads a WAVE from the data source, automatically freeing |
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190 |
* that source if 'freesrc' is non-zero. For example, to load a WAVE file, |
|
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* you could do: |
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* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); |
|
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* |
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* If this function succeeds, it returns the given SDL_AudioSpec, |
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* filled with the audio data format of the wave data, and sets |
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* 'audio_buf' to a malloc()'d buffer containing the audio data, |
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* and sets 'audio_len' to the length of that audio buffer, in bytes. |
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* You need to free the audio buffer with SDL_FreeWAV() when you are |
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* done with it. |
|
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* |
|
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* This function returns NULL and sets the SDL error message if the |
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* wave file cannot be opened, uses an unknown data format, or is |
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* corrupt. Currently raw and MS-ADPCM WAVE files are supported. |
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*/ |
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extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, |
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int freesrc, |
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SDL_AudioSpec * spec, |
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Uint8 ** audio_buf, |
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Uint32 * audio_len); |
0 | 210 |
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/* Compatibility convenience function -- loads a WAV from a file */ |
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#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ |
|
213 |
SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) |
|
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215 |
/* |
|
216 |
* This function frees data previously allocated with SDL_LoadWAV_RW() |
|
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*/ |
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extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); |
0 | 219 |
|
220 |
/* |
|
221 |
* This function takes a source format and rate and a destination format |
|
222 |
* and rate, and initializes the 'cvt' structure with information needed |
|
223 |
* by SDL_ConvertAudio() to convert a buffer of audio data from one format |
|
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* to the other. |
|
225 |
* This function returns 0, or -1 if there was an error. |
|
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*/ |
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extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, |
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Uint16 src_format, |
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Uint8 src_channels, |
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int src_rate, |
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Uint16 dst_format, |
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Uint8 dst_channels, |
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int dst_rate); |
0 | 234 |
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/* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(), |
|
236 |
* created an audio buffer cvt->buf, and filled it with cvt->len bytes of |
|
237 |
* audio data in the source format, this function will convert it in-place |
|
238 |
* to the desired format. |
|
239 |
* The data conversion may expand the size of the audio data, so the buffer |
|
240 |
* cvt->buf should be allocated after the cvt structure is initialized by |
|
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* SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long. |
|
242 |
*/ |
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extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); |
0 | 244 |
|
245 |
/* |
|
246 |
* This takes two audio buffers of the playing audio format and mixes |
|
247 |
* them, performing addition, volume adjustment, and overflow clipping. |
|
248 |
* The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME |
|
249 |
* for full audio volume. Note this does not change hardware volume. |
|
250 |
* This is provided for convenience -- you can mix your own audio data. |
|
251 |
*/ |
|
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#define SDL_MIX_MAXVOLUME 128 |
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extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, |
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Uint32 len, int volume); |
0 | 255 |
|
256 |
/* |
|
257 |
* The lock manipulated by these functions protects the callback function. |
|
258 |
* During a LockAudio/UnlockAudio pair, you can be guaranteed that the |
|
259 |
* callback function is not running. Do not call these from the callback |
|
260 |
* function or you will cause deadlock. |
|
261 |
*/ |
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extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
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extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
0 | 264 |
|
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/* |
|
266 |
* This function shuts down audio processing and closes the audio device. |
|
267 |
*/ |
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extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
0 | 269 |
|
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||
271 |
/* Ends C function definitions when using C++ */ |
|
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#ifdef __cplusplus |
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/* *INDENT-OFF* */ |
0 | 274 |
} |
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/* *INDENT-ON* */ |
0 | 276 |
#endif |
277 |
#include "close_code.h" |
|
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||
279 |
#endif /* _SDL_audio_h */ |
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|
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/* vi: set ts=4 sw=4 expandtab: */ |